Voice over Internet Protocol (VoIP) is the transport of voice traffic using the Internet Protocol (IP). In the mobile world, VoIP means using a packet-switched (PS) service for transport of Internet Protocol (IP) packets which contain, e.g., Adaptive Multi-Rate (AMR) codec speech frames for voice mobile phone calls. A packet-switched connection is often simply referred to as a data connection.
In packet-switched networks, the message is broken into packets, each of which can take a different route to the destination where the packets are recompiled into the original message. The packet switched (PS) service utilized for VoIP can be, for example, GPRS (General Packet Radio Service), EDGE (Enhanced Data Rates for Global Evolution), or WCDMA (Wideband Code Division Multiple Access). Each of these example services happen to be built upon the Global System for Mobile communications (GSM), a second generation (“2G”) digital radio access technology originally developed for Europe. GSM was enhanced in 2.5G to include technologies such as GPRS. The third generation (3G) comprises mobile telephone technologies covered by the International Telecommunications Union (ITU) IMT-2000 family. The Third Generation Partnership Project (3GPP) is a group of international standards bodies, operators, and vendors working toward standardizing WCDMA-based members of the IMT-2000.
EDGE (sometimes referred to as Enhanced GPRS (EGPRS)) is a 3G technology that delivers broadband-like data speeds to mobile devices. EDGE allows consumers to connect to the Internet and send and receive data, including digital images, web pages and photographs, three times faster than possible with an ordinary GSM/GPRS network. EDGE enables GSM operators to offer higher-speed mobile-data access, serve more mobile-data customers, and free up GSM network capacity to accommodate additional voice traffic. EDGE uses the same TDMA (Time Division Multiple Access) frame structure, logical channels, and 200 kHz carrier bandwidth as GSM networks, which allows existing cell plans to remain intact.
In EDGE technology, a base transceiver station (BTS) communicates with a mobile station (e.g., a cell phone, mobile terminal or the like, including computers such as laptops with mobile termination). The base transceiver station (BTS) typically has plural transceivers (TRX). A time division multiple access (TDMA) radio communication system like GSM, GPRS, and EDGE divides the time space into time slots on a particular radio frequency. Time slots are grouped into frames, with users being assigned one or more time slots. In packet-switched TDMA, even though one user might be assigned one or more time slots, other users may use the same time slot(s). So a time slot scheduler is needed to ensure that the time slots are allocated properly and efficiently.
EDGE offers nine different Modulation and Coding Schemes (MCSs): MCS 1 through MCS9. Lower coding schemes (e.g., MCS1-MCS2) deliver a more reliable but slower bit rate and are suitable for less optimal radio conditions. Higher coding schemes (e.g., MCS8-MCS9) deliver a much higher bit rate, but require better radio conditions. Link Quality Control (LQC) selects which MCS to use in each particular situation based on the current radio conditions.
In EDGE, the LQC selects a MCS for radio link control (RLC) data blocks for each temporary block flow (TBF). A TBF is a logical connection between a mobile station (MS) and a packet control unit (PCU). The PCU is usually (but not necessarily) located in the radio access network, e.g., in the base station controller (BSC). A TBF is used for either uplink or downlink transfer of GPRS packet data. The actual packet transfer is made on physical data radio channels (PDCHs). The bit rate for a TBF is thus effectively selected by selecting a MCS, and changing the MCS for a TBF changes its bit rate.
Adaptive Multi-rate (AMR) speech frames contain speech, typically 20 milliseconds of speech, encoded by an AMR codec. Voice encoder, vocoder, and codec are used interchangeably and refer to encoding speech/voice into a compressed digital format. An AMR codec supports unequal bit-error detection and protection (UED/UEP). The UEP/UED mechanisms allow more efficient transmission of speech over a lossy network by sorting the bits into perceptually more and less sensitive classes. A frame is only declared damaged and not delivered if there are one or more bit errors found in the most sensitive bits. On the other hand, speech quality is still deemed acceptable if the speech frame is delivered with one or more bit errors in the less sensitive bits, based on human aural perception. In VoIP the AMR codec does not provide any redundancy for channel coding. The AMR codec only produces speech output and, instead, it is the EDGE coding schemes that are used for redundancy. EDGE can provide an error-free bearer with RLC acknowledged mode.
Another benefit of AMR is adaptive rate adaptation for switching smoothly between codec modes on-the-fly. A large number of AMR codec modes may be used with varying bit rates and resulting voice quality. An AMR codec may include multiple narrowband codec modes: 12.2, 10.2, 7.95, 7.4, 6.7, 5.9, 5.5 and 4.75 kbit/s. Even a wideband (WB) mode AMR WB at 12.65 kbit/s is available.
Traffic is increasing in GSM networks around the world and more and more networks are becoming interference limited. Hence, spectrum and interference levels are limiting factors for how much traffic that can be sent. One way to increase traffic in such a situation is to reduce the amount of energy needed per user which can e.g. be achieved by power control, improved receivers, more efficient coding or directing the energy using adaptive antennas.
Existing power control algorithms typically adapt an output power to a radio quality of a connection, and sometimes also in combination with a service/bearer (in UTRAN) which can be used. In situations where a coding scheme, and/or code selection, can be modified due to limited amount of data to transmit, then excessive power is used.
As an example in GPRS/EDGE data packets are queued in the PCU buffer before transferred over the air interface. Whenever the PCU buffer runs empty the last amount of data will be transmitted using a more robust MCS that can carry the remaining amount of data e.g. using for example MCS-5 instead of MCS-9, in the case when the LQC recommends a higher MCS than needed to fit the last amount of data into one radio block. Such a procedure is mentioned in the standard, 3GPP TS 44.060 V7.7.0 (2006-12) “Radio Link Control/Medium Access Control (RLC/MAC) protocol (Release 7)”.
How often the PCU buffer runs empty during an ongoing session depends on the application type. Using excessive power has a negative impact on the system capacity and buffer under-runs happen all the time with VoIP conversations.